Guys, does anyone know if DO has a block on using PBX systems, particularly Freepbx? I’m going on two days trying to register a trunk with a number from Telnyx and I can’t seem to get it to work on Freepbx. I’ve set up several droplets and nothing works. I keep getting either dropped calles or Number not in service. I’ve contacted Telnyx and they’ve advised on working on their end. The only thing I can think would be a server related issue since DO has emails blocked. Also, I’ve set up droplets with this: FreePBX® | DigitalOcean Marketplace 1-Click App Thanks in advance.
I am the maintainer of that 1-Click image and can vouch for it, also know of many installations using it currently. I can almost guarantee there’s no general block on PBX traffic to/from Digital Ocean.
You will need to investigate the FreePBX firewall on your end and also add your IP at Telnyx.
Thanks. Telnyx support says that a signal is being sent to the pbx but, the pbx ip number returns number unnavilable. At this point I’m stuck. Also, I haven’t been able to register the Polycom IP450 phone to Freepbx. thanks
would anyone have an idea why I can’t get this pbx to register?
This sounds like the SIP traffic is coming in but you don’t have a correct Inbound Route for it. Or possibly you don’t have the trunk set up correctly and Asterisk is rejecting it. Use PJSIP trunks and set the SIP server to sip.telnyx.com. Look at the Asterisk log to see what’s going on.
That’s exactly how I have it. How do I Look at the Asterisk log to see what’s going on?
Hi, I got it dial out but I’m still getting Not in service going in… Any idea?
The Asterisk log can be viewed in the GUI at Reports → Asterisk Logfiles, or you can access it directly at
/var/log/asterisk/full
If the log doesn’t show enough detail, at the Asterisk command prompt type
pjsip set logger on
make another failing incoming call and the new log will include a SIP trace.
If you have trouble interpreting the log, paste the relevant section at pastebin.com and post the link here.
If nothing gets logged on an attempted call, report what, if anything, appears in sngrep for an attempt.
Hi, this is the last thing I got:
res_pjsip/pjsip_options.c: Contact Telnyx_ip/sip:sip.telnyx.com:5060 is now Reachable. RTT: 84.944 msec
Assuming that the timestamp is earlier than when you attempted a call, nothing got added to the log when you tried.
Under SIP Trunking, what Connection Type are you using at Telnyx (Credentials, IP Address, etc.)?
Under My Numbers, do you have the number you are calling pointing to the proper Connection?
I am using credentials and the only number I’m working with is pointed to the only connection
In Reports → Asterisk Info → Registries, does the Telnyx trunk show Registered?
In Telnyx Reports → Reporting, get a detailed report of the time period. What, if anything, shows for the attempted calls?
At the Asterisk command prompt, type
pjsip set logger on
and attempt an inbound call. What, if anything, appears in the Asterisk log?
If nothing, run sngrep and report what appears there.
In Registries: No objects found.
Confirm that your Telnyx trunk has Authentication Outbound and Registration Send. Client URI and Server URI should probably be blank (explain if not). The settings related to retries and auth rejection should be left at defaults.
Search the Asterisk log for any entries related to registration failure.
Now I don’t follow. sorry
If you are using Credentials on Telnyx, the PBX must register to them in order to receive calls. This is apparently not happening because you said Registries is empty. So, most likely, the trunk is not set up correctly. If you don’t understand the settings, please post screenshots of both pjsip General and Advanced tabs for your trunk. (Mask your username and secret.)
Never mind. Thank you very much for all your help. I changed the extension number and it registered.
Actually I have a question. Although is working, the calls seem to be going into voice mail. How can I get the calls to ring the phone extension instead? TY
Test that the extension is working correctly by calling from a second extension (use a softphone or SIP app if you don’t have another device).
If so, temporarily set up a default (any/any) Inbound Route to the extension in question.
If you still have trouble, with pjsip logger on, paste the Asterisk log for the failed call (that went to voicemail instead of ringing the extension) at pastebin.com and post the link here.
Inbound route> DID to Extension…